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gst/rtp/: Ported mulaw and alaw payloaders to use new base class

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Original commit message from CVS:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
This commit is contained in:
Philippe Kalaf 2007-03-14 22:21:26 +00:00
parent c209a3f894
commit 1be3219c70
5 changed files with 33 additions and 280 deletions
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7
ChangeLog
View file

@ -1,3 +1,10 @@
2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmapay.h:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtppcmupay.h:
Ported mulaw and alaw payloaders to use new base class
2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
* po/af.po:

142
gst/rtp/gstrtppcmapay.c
View file

@ -59,17 +59,9 @@ static GstStaticPadTemplate gst_rtp_pcma_pay_src_template =
static gboolean gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static void gst_rtp_pcma_pay_finalize (GObject * object);
GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
/* The lower limit for number of octet to put in one packet
* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
* to to 10msec (see RFC3551) */
#define GST_RTP_PCMA_MIN_PTIME_OCTETS 80
GST_BOILERPLATE (GstRtpPmcaPay, gst_rtp_pcma_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcma_pay_base_init (gpointer klass)
@ -95,30 +87,24 @@ gst_rtp_pcma_pay_class_init (GstRtpPmcaPayClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_pcma_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcma_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_pcma_pay_handle_buffer;
}
static void
gst_rtp_pcma_pay_init (GstRtpPmcaPay * rtppcmapay, GstRtpPmcaPayClass * klass)
{
rtppcmapay->adapter = gst_adapter_new ();
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmapay);
GST_BASE_RTP_PAYLOAD (rtppcmapay)->clock_rate = 8000;
}
static void
gst_rtp_pcma_pay_finalize (GObject * object)
{
GstRtpPmcaPay *rtppcmapay;
/* tell basertpaudiopayload that this is a sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
rtppcmapay = GST_RTP_PCMA_PAY (object);
g_object_unref (rtppcmapay->adapter);
rtppcmapay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
/* octet-per-sample is 1 for PCM */
gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
@ -132,114 +118,6 @@ gst_rtp_pcma_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
return TRUE;
}
static GstFlowReturn
gst_rtp_pcma_pay_flush (GstRtpPmcaPay * rtppcmapay, guint32 clock_rate)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = GST_RTP_PCMA_MIN_PTIME_OCTETS;
if (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime > 0) {
/* calculate octet count with:
maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
maxptime_octets =
gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmapay)->max_ptime,
clock_rate, GST_SECOND);
}
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. */
avail = gst_adapter_available (rtppcmapay->adapter);
ret = GST_FLOW_OK;
while (avail >= minptime_octets) {
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* fill one MTU or all available bytes */
payload_len =
MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmapay), maxptime_octets),
avail);
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtppcmapay));
payload = gst_rtp_buffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtppcmapay->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtppcmapay->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtppcmapay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmapay), outbuf);
/* increase count (in ts) of data pushed to basertppayload */
rtppcmapay->first_ts +=
gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
/* store amount of unpushed data (in ts) */
rtppcmapay->duration =
gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
}
return ret;
}
static GstFlowReturn
gst_rtp_pcma_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpPmcaPay *rtppcmapay;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
guint32 clock_rate;
rtppcmapay = GST_RTP_PCMA_PAY (basepayload);
clock_rate = basepayload->clock_rate;
size = GST_BUFFER_SIZE (buffer);
duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
avail = gst_adapter_available (rtppcmapay->adapter);
if (avail == 0) {
rtppcmapay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmapay->duration = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtppcmapay->duration + duration)) {
ret = gst_rtp_pcma_pay_flush (rtppcmapay, clock_rate);
/* note: first_ts and duration updated in ...pay_flush() */
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtppcmapay->adapter, buffer);
rtppcmapay->duration += duration;
return ret;
}
gboolean
gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin)
{

11
gst/rtp/gstrtppcmapay.h
View file

@ -17,8 +17,7 @@
#define __GST_RTP_PCMA_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
@ -38,16 +37,12 @@ typedef struct _GstRtpPmcaPayClass GstRtpPmcaPayClass;
struct _GstRtpPmcaPay
{
GstBaseRTPPayload payload;
GstAdapter *adapter;
GstClockTime first_ts;
GstClockTime duration;
GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpPmcaPayClass
{
GstBaseRTPPayloadClass parent_class;
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_pcma_pay_plugin_init (GstPlugin * plugin);

142
gst/rtp/gstrtppcmupay.c
View file

@ -59,17 +59,9 @@ static GstStaticPadTemplate gst_rtp_pcmu_pay_src_template =
static gboolean gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
static void gst_rtp_pcmu_pay_finalize (GObject * object);
GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
/* The lower limit for number of octet to put in one packet
* (clock-rate=8000, octet-per-sample=1). The default 80 is equal
* to to 10msec (see RFC3551) */
#define GST_RTP_PCMU_MIN_PTIME_OCTETS 80
GST_BOILERPLATE (GstRtpPcmuPay, gst_rtp_pcmu_pay, GstBaseRTPAudioPayload,
GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
static void
gst_rtp_pcmu_pay_base_init (gpointer klass)
@ -95,30 +87,24 @@ gst_rtp_pcmu_pay_class_init (GstRtpPcmuPayClass * klass)
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_rtp_pcmu_pay_finalize;
gstbasertppayload_class->set_caps = gst_rtp_pcmu_pay_setcaps;
gstbasertppayload_class->handle_buffer = gst_rtp_pcmu_pay_handle_buffer;
}
static void
gst_rtp_pcmu_pay_init (GstRtpPcmuPay * rtppcmupay, GstRtpPcmuPayClass * klass)
{
rtppcmupay->adapter = gst_adapter_new ();
GstBaseRTPAudioPayload *basertpaudiopayload;
basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtppcmupay);
GST_BASE_RTP_PAYLOAD (rtppcmupay)->clock_rate = 8000;
}
static void
gst_rtp_pcmu_pay_finalize (GObject * object)
{
GstRtpPcmuPay *rtppcmupay;
/* tell basertpaudiopayload that this is a sample based codec */
gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
rtppcmupay = GST_RTP_PCMU_PAY (object);
g_object_unref (rtppcmupay->adapter);
rtppcmupay->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
/* octet-per-sample is 1 for PCM */
gst_base_rtp_audio_payload_set_sample_options (basertpaudiopayload, 1);
}
static gboolean
@ -132,114 +118,6 @@ gst_rtp_pcmu_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
return TRUE;
}
static GstFlowReturn
gst_rtp_pcmu_pay_flush (GstRtpPcmuPay * rtppcmupay, guint32 clock_rate)
{
guint avail;
GstBuffer *outbuf;
GstFlowReturn ret;
guint maxptime_octets = G_MAXUINT;
guint minptime_octets = GST_RTP_PCMU_MIN_PTIME_OCTETS;
if (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime > 0) {
/* calculate octet count with:
maxptime-nsec * samples-per-sec / nsecs-per-sec * octets-per-sample */
maxptime_octets =
gst_util_uint64_scale_int (GST_BASE_RTP_PAYLOAD (rtppcmupay)->max_ptime,
clock_rate, GST_SECOND);
}
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. */
avail = gst_adapter_available (rtppcmupay->adapter);
ret = GST_FLOW_OK;
while (avail >= minptime_octets) {
guint8 *payload;
guint8 *data;
guint payload_len;
guint packet_len;
/* fill one MTU or all available bytes */
payload_len =
MIN (MIN (GST_BASE_RTP_PAYLOAD_MTU (rtppcmupay), maxptime_octets),
avail);
/* this will be the total lenght of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (payload_len, 0, 0);
/* create buffer to hold the payload */
outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
/* copy payload */
gst_rtp_buffer_set_payload_type (outbuf,
GST_BASE_RTP_PAYLOAD_PT (rtppcmupay));
payload = gst_rtp_buffer_get_payload (outbuf);
data = (guint8 *) gst_adapter_peek (rtppcmupay->adapter, payload_len);
memcpy (payload, data, payload_len);
gst_adapter_flush (rtppcmupay->adapter, payload_len);
avail -= payload_len;
GST_BUFFER_TIMESTAMP (outbuf) = rtppcmupay->first_ts;
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtppcmupay), outbuf);
/* increase count (in ts) of data pushed to basertppayload */
rtppcmupay->first_ts +=
gst_util_uint64_scale_int (payload_len, GST_SECOND, clock_rate);
/* store amount of unpushed data (in ts) */
rtppcmupay->duration =
gst_util_uint64_scale_int (avail, GST_SECOND, clock_rate);
}
return ret;
}
static GstFlowReturn
gst_rtp_pcmu_pay_handle_buffer (GstBaseRTPPayload * basepayload,
GstBuffer * buffer)
{
GstRtpPcmuPay *rtppcmupay;
guint size, packet_len, avail;
GstFlowReturn ret;
GstClockTime duration;
guint32 clock_rate;
rtppcmupay = GST_RTP_PCMU_PAY (basepayload);
clock_rate = basepayload->clock_rate;
size = GST_BUFFER_SIZE (buffer);
duration = gst_util_uint64_scale_int (size, GST_SECOND, clock_rate);
avail = gst_adapter_available (rtppcmupay->adapter);
if (avail == 0) {
rtppcmupay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
rtppcmupay->duration = 0;
}
/* get packet length of data and see if we exceeded MTU. */
packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
if (gst_basertppayload_is_filled (basepayload,
packet_len, rtppcmupay->duration + duration)) {
/* note: first_ts and duration updated in ...pay_flush() */
ret = gst_rtp_pcmu_pay_flush (rtppcmupay, clock_rate);
} else {
ret = GST_FLOW_OK;
}
gst_adapter_push (rtppcmupay->adapter, buffer);
rtppcmupay->duration += duration;
return ret;
}
gboolean
gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin)
{

11
gst/rtp/gstrtppcmupay.h
View file

@ -17,8 +17,7 @@
#define __GST_RTP_PCMU_PAY_H__
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstbasertpaudiopayload.h>
G_BEGIN_DECLS
@ -38,16 +37,12 @@ typedef struct _GstRtpPcmuPayClass GstRtpPcmuPayClass;
struct _GstRtpPcmuPay
{
GstBaseRTPPayload payload;
GstAdapter *adapter;
GstClockTime first_ts;
GstClockTime duration;
GstBaseRTPAudioPayload audiopayload;
};
struct _GstRtpPcmuPayClass
{
GstBaseRTPPayloadClass parent_class;
GstBaseRTPAudioPayloadClass parent_class;
};
gboolean gst_rtp_pcmu_pay_plugin_init (GstPlugin * plugin);